Demo Data from SIGIR 2026 / reviewers in the wild / expert
Wenbin Jiang 0003
dblp:96/5583-3
· DBLP profile ↗
6ranked-venue papers
6as first author
6since 2021 · last 2025
0000-0002-4063-8952ORCID · verified
Domains — the database's venue-derived domains; a paper can count in several
Artificial intelligence and machine learning · 4 · 4 first-author · 4 since 2021Graphics, computer vision, multimedia, augmented reality and games · 4 · 4 first-author · 4 since 2021
Expertise — from the expertise taxonomy: the topics of the expert's papers under the CCF categories. A weight counts papers with recency: 1 for a paper about the topic, 0.3 when the topic is its context, halved every five years.
| Computer graphics and multimedia
2 papers |
Audio and music processing · 68% Virtual and augmented reality · 32% |
Topics — the 2 heaviest of 2, each with the papers that count most for it
| Topic | Weight | Papers | Last | Evidence papers |
|---|---|---|---|---|
Audio and music processing
speech enhancement |
1.4 | 2 | 2024 | Unsupervised Speech Enhancement Using Optimal Transport and Speech Presence Probability · IEEE ACM Trans. Audio Speech Lang. Process. 2024 Speech Enhancement With Integration of Neural Homomorphic Synthesis and Spectral Masking · IEEE ACM Trans. Audio Speech Lang. Process. 2023 |
Virtual and augmented reality › auditory perception › psychoacoustics
frequency masking |
0.7 | 1 | 2023 | Speech Enhancement With Integration of Neural Homomorphic Synthesis and Spectral Masking · IEEE ACM Trans. Audio Speech Lang. Process. 2023 |
Methods — techniques the papers use, named apart from their topics
speech presence probability · 0.8optimal transport · 0.8multi-resolution STFT loss · 0.8shared-encoder multi-decoder network · 0.7complex spectral masking · 0.7cepstrum liftering · 0.7
| Year | Publication | Venue | Position |
|---|---|---|---|
| 2025 | MOS-GAN: Mean Opinion Score GAN for Unsupervised Speech Enhancement
Wenbin Jiang 0003, Fei Wen 0005, Kai Yu 0004 |
IEEE Signal Process. Lett. | 1 |
| 2024 | Unsupervised Speech Enhancement Using Optimal Transport and Speech Presence ProbabilityabstractSpeech enhancement models based on deep learning are typically trained in a supervised manner, requiring a substantial amount of paired noisy-to-clean speech data for training. However, synthetically generated training data can only capture a limited range of realistic environments, and it is often challenging or even impractical to gather real-world pairs of noisy and ground-truth clean speech. To overcome this limitation, we propose an unsupervised learning approach for speech enhancement that eliminates the need for paired noisy-to-clean training data. Specifically, our method utilizes the optimal transport criterion to train the speech enhancement model in an unsupervised manner. It employs a fidelity loss based on noisy speech and a distribution divergence loss to minimize the difference between the distribution of the model's output and that of unpaired clean speech. Further, we use the speech presence probability as an additional optimization objective and incorporate the short-time Fourier transform (STFT) domain loss as an extra term for the unsupervised learning loss. We also apply the multi-resolution STFT loss as the validation loss to enhance the stability of the training process and improve the algorithm's performance. Experimental results on the VCTK + DEMAND benchmark demonstrate that the proposed method achieves competitive performance compared to the supervised methods. Furthermore, the speech recognition results on the CHiME4 benchmark show the superiority of the proposed method over its supervised counterpart. Wenbin Jiang 0003, Kai Yu 0004, Fei Wen 0005 |
IEEE ACM Trans. Audio Speech Lang. Process. | 1 |
| 2023 | UnSE: Unsupervised Speech Enhancement Using Optimal Transport
Wenbin Jiang 0003, Fei Wen 0005, Kai Yu 0004 |
INTERSPEECH | 1 |
| 2023 | Speech Enhancement With Integration of Neural Homomorphic Synthesis and Spectral MaskingabstractSpeech enhancement refers to suppressing the background noise to improve the perceptual quality and intelligibility of the observed noisy speech. Recently, speech enhancement algorithms based on deep neural networks (DNNs) have replaced traditional algorithms based on statistical signal processing and have become mainstream in the research field. However, most DNN-based speech enhancement methods commonly operate on the frequency domain and do not use the speech production model, which makes the models prone to under-suppress the noise or over-suppress the speech. To address the shortcoming, we propose a novel speech enhancement method integrating neural homomorphic synthesis and complex spectral masking. Specifically, we use a shared-encoder and multi-decoder neural network architecture. For the neural homomorphic synthesis branch, the speech signal is separated into excitation and vocal tract components through liftering the cepstrum, two DNN decoders are applied to estimate the target components independently, and the denoised speech is synthesized by the estimated minimum-phase signal and the noisy phase. For the spectral masking branch, another DNN decoder is adopted to estimate the complex mask of the target spectrum, and the denoised speech spectrum is obtained by masking the noisy spectrum. The two branches respectively estimate speech signals, and the final enhanced speech is obtained by merging the two branches of estimated speech. Experimental results on two popular datasets show that the proposed method achieves state-of-the-art level performance, with only 920 K model parameters. Wenbin Jiang 0003, Kai Yu 0004 |
IEEE ACM Trans. Audio Speech Lang. Process. | 1 |
| 2022 | Speech Enhancement with Neural Homomorphic SynthesisabstractMost deep learning-based speech enhancement methods operate directly on time-frequency representations or learned features without making use of the model of speech production. This work proposes a new speech enhancement method based on neural homomorphic synthesis. The speech signal is firstly decomposed into excitation and vocal tract with complex cepstrum analysis. Then, two complex-valued neural networks are applied to estimate the target complex spectrum of the decomposed components. Finally, the time-domain speech signal is synthesized from the estimated excitation and vocal tract. Furthermore, we investigated numerous loss functions and found that the multi-resolution STFT loss, commonly used in the TTS vocoder, benefits speech enhancement. Experimental results demonstrate that the proposed method outperforms existing state-of-the-art complex-valued neural network-based methods in terms of both PESQ and eSTOI. Wenbin Jiang 0003, Kai Yu 0004, Fei Wen 0005 |
ICASSP | 1 |
| 2022 | Efficient Speech Enhancement with Neural Homomorphic Synthesis
Wenbin Jiang 0003, Tao Liu 0068, Kai Yu 0004 |
INTERSPEECH | 1 |